Abstract: A headset adapted to transmitting an outgoing audio signal (Tx) and comprising a voice microphone and at least one ambient microphone. The headset comprises an ambient noise reduction block (ANR), which is adapted to reduce the level of ambient noise in the outgoing audio signal (Tx). The ambient noise reduction block (ANR) includes the measuring the levels of the voice microphone signal (X) and ambient microphone signal (Y) to estimate a characteristic constant level drop between the voice microphone signal (X) and the ambient microphone signal (Y), which is characteristic for the headset user talking. A time-varying filter passes the voice microphone signal (X), when the level difference is larger than characteristic constant level drop (CLD) and attenuates the voice microphone signal (X), if the level difference is below characteristic constant level drop.

Abstract: The present invention relates to a microphone apparatus (30) with a main beamformer (31) that provides a main output audio signal (SM) as a beamformed signal by applying a main weight vector (BM) to a main input vector (MM). A main beamformer controller (32) repeatedly determines a main steering vector (dM) and adaptively determines the main weight vector (BM) in dependence on the main steering vector (dM) and the main input vector (MM) to increase the relative amount of voice sound (V) from the user (6) in the main output audio signal (SM).

Abstract: A headset adapted to transmitting an outgoing audio signal (Tx) and comprising a voice microphone and at least one ambient microphone. The headset comprises an ambient noise reduction block (ANR), which is adapted to reduce the level of ambient noise in the outgoing audio signal (Tx). The ambient noise reduction block (ANR) includes the measuring the levels of the voice microphone signal (X) and ambient microphone signal (Y) to estimate a characteristic constant level drop between the voice microphone signal (X) and the ambient microphone signal (Y), which is characteristic for the headset user talking. A time-varying filter passes the voice microphone signal (X), when the level difference is larger than characteristic constant level drop (CLD) and attenuates the voice microphone signal (X), if the level difference is below characteristic constant level drop.

Abstract: The present invention relates to a microphone apparatus (30) with a main beamformer (31) that provides a main output audio signal (SM) as a beamformed signal by applying a main weight vector (BM) to a main input vector (MM). A main beamformer controller (32) repeatedly determines a main steering vector (dM) and adaptively determines the main weight vector (BM) in dependence on the main steering vector (dM) and the main input vector (MM) to increase the relative amount of voice sound (V) from the user (6) in the main output audio signal (SM).

Abstract: The present invention relates to a microphone apparatus (10) with a main beamformer (F, BF) that provides a directional audio output (SF) by combining microphone signals (X, Y) from multiple microphones (11, 12). The quality of beamformed microphone signals normally depends on the individual microphones having equal sensitivity characteristics across the used frequency range. The invention enables automatic adaptation of the main beamformer (F, BF) to variations in microphone sensitivity and to changes in the alignment of the microphone apparatus (10) with respect to the user's mouth (7).

Abstract: The present invention relates to a microphone apparatus (10) with a main beamformer (F, BF) that provides a directional audio output (SF) by combining microphone signals (X, Y) from multiple microphones (11, 12). The quality of beamformed microphone signals normally depends on the individual microphones having equal sensitivity characteristics across the used frequency range. The invention enables automatic adaptation of the main beamformer (F, BF) to variations in microphone sensitivity and to changes in the alignment of the microphone apparatus (10) with respect to the user's mouth (7).

Abstract: The present invention relates to a desktop speakerphone constructed to make the desktop speakerphone less space-consuming while having improved audio qualities. The desktop speakerphone preferably has two microphone clusters 6, 7 mounted at the upper side of the housing 2 closer towards respective longitudinal ends 8 of the latter, so that each microphone cluster 6, 7 can receive voice sound Av from one or more of the users. Each microphone cluster 6, 7 preferably comprises three pressure microphones 10, 11, 12. Furthermore, within each microphone cluster 6, 7, the second and third sound inlets 14, 15 are preferably arranged symmetrically on opposite sides of the respective median plane 18. Within each microphone cluster 6, 7, the relative arrangement of the three sound inlets 13, 14, 15 defines a respective microphone axis 9, 19 for each of the microphone pairs 10, 11, 10, 12.

Abstract: The present invention relates to a desktop speakerphone array processor for microphones which produces high quality sound. The microphone array has a front direction defined by the line of sight from a microphone inlet of the second microphone towards a microphone inlet of the first microphone, the array processor being connected to receive a front microphone signal from the first microphone, a rear microphone signal from the second microphone and an audio output signal representing a speaker sound emitted from a sound driver arranged near the first and second microphones and in the rearwards hemisphere with respect to the front direction of the microphone array, the array processor being configured to provide a first array signal having a first directivity pattern with a main lobe oriented in the front direction of the microphone array in dependence on the front microphone signal, the rear microphone signal and the audio output signal.

Abstract: The present invention relates to a desktop speakerphone constructed to make the desktop speakerphone less space-consuming while having improved audio qualities. The desktop speakerphone preferably has two microphone clusters 6, 7 mounted at the upper side of the housing 2 closer towards respective longitudinal ends 8 of the latter, so that each microphone cluster 6, 7 can receive voice sound Av from one or more of the users. Each microphone cluster 6, 7 preferably comprises three pressure microphones 10, 11, 12. Furthermore, within each microphone cluster 6, 7, the second and third sound inlets 14, 15 are preferably arranged symmetrically on opposite sides of the respective median plane 18. Within each microphone cluster 6, 7, the relative arrangement of the three sound inlets 13, 14, 15 defines a respective microphone axis 9, 19 for each of the microphone pairs 10, 11, 10, 12.

Abstract: The present invention relates to a desktop speakerphone array processor for microphones which produces high quality sound. The microphone array has a front direction defined by the line of sight from a microphone inlet of the second microphone towards a microphone inlet of the first microphone, the array processor being connected to receive a front microphone signal from the first microphone, a rear microphone signal from the second microphone and an audio output signal representing a speaker sound emitted from a sound driver arranged near the first and second microphones and in the rearwards hemisphere with respect to the front direction of the microphone array, the array processor being configured to provide a first array signal having a first directivity pattern with a main lobe oriented in the front direction of the microphone array in dependence on the front microphone signal, the rear microphone signal and the audio output signal.

Abstract: The present invention relates to a desktop speakerphone constructed to make the desktop speakerphone less space-consuming while having improved audio qualities. The desktop speakerphone preferably has two microphone clusters 6, 7 mounted at the upper side of the housing 2 closer towards respective longitudinal ends 8 of the latter, so that each microphone cluster 6, 7 can receive voice sound Av from one or more of the users. Each microphone cluster 6, 7 preferably comprises three pressure microphones 10, 11, 12. Furthermore, within each microphone cluster 6, 7, the second and third sound inlets 14, 15 are preferably arranged symmetrically on opposite sides of the respective median plane 18. Within each microphone cluster 6, 7, the relative arrangement of the three sound inlets 13, 14, 15 defines a respective microphone axis 9, 19 for each of the microphone pairs 10, 11, 10, 12.